Awesome. Feel free to submit a PR when you're ready.
I want to do the SSL release as soon as possible but I don't think it'll
happen until March or so..
Romain
2017-02-18 15:30 GMT-06:00 Mark E <[email protected]>:
>
> Great tips, much appreciated. I think I have the basics working -
> woohoooo!
>
> I need to put this code out on an Internet accessible dev box (get it off
> my localhost) so I can do some testing with more than one caller. Which I
> can do. BUT, one small problem: The current version of liquidsoap is 1.2.1
> and it doesn't appear to have support for the new input.harbor.ssl inputs.
> This presents a problem because Chrome refuses to connect to insecure
> non-SSL Web sockets unless it's on a localhost network (127) network.
>
> Can you all please finalize the liquidsoap build for the opam package so
> that it has the new SSL features? (RE: Github #376 linked below - January
> 7, 2017 - "SSL support for harbor", looks like it may be 99% ready. )
>
> https://github.com/savonet/liquidsoap/pull/376
> <http://h5ks.mjt.lu/lnk/AEAAHpZTiiEAAAAAAAAAAGgH3uUAASJ-8QAAAAAAAAgArwBYqLzevEDLj7I6TJaVrbRJp9rrAgAHnU0/1/wMJ_1W7aJmah87N_NacWeQ/aHR0cHM6Ly9naXRodWIuY29tL3Nhdm9uZXQvbGlxdWlkc29hcC9wdWxsLzM3Ng>
>
> SSL would a really great enabling feature so that people that need to can
> put liquidsoap on a non-localhost network system (e.g. have a totally
> web-based web-hosted system).
>
> Anyway when I finish testing and tweaking my Webcaster mods I can share.
>
> BTW: If you guys had a donate button somewhere I'd be more than happy to
> send some bux to help get the SSL feature rolled out!
>
> Mark
>
>
>
>
> On 02/18/2017 09:54 AM, Romain Beauxis wrote:
>
> Hi,
>
> Thanks for your interest! This seems like a nice feature to experiment
> with!
>
> The code is a bit hairy but so are the various JS Audio APIs.. I'd suggest
> that you look at the microphone code, since a WebRTC call in seems pretty
> similar to streaming from the microphone..
>
> The code for the model is here: https://github.com/
> webcast/webcaster/blob/master/src/models/microphone.coffee
> <http://h5ks.mjt.lu/lnk/AEAAHpZTiiEAAAAAAAAAAGgH3uUAASJ-8QAAAAAAAAgArwBYqLzevEDLj7I6TJaVrbRJp9rrAgAHnU0/2/rYTh_6Oi50UEeohpRKcMnA/aHR0cHM6Ly9naXRodWIuY29tL3dlYmNhc3Qvd2ViY2FzdGVyL2Jsb2IvbWFzdGVyL3NyYy9tb2RlbHMvbWljcm9waG9uZS5jb2ZmZWU>
>
> The class inherits a @destination audio node which is where you want to
> plug stuff. @destination is filled in after calling @prepare(). Then in
> createSource, the audio source is asynchronously created.
>
> A quick path to experiment could be to simply hack on this class and use
> the existing microphone UI until you have it working as you wish and then
> split it into a new source in the webcaster UI.
>
> Hope that makes sense and can help. Remember to rebuild your coffee script
> stuff after doing your changes! `make` would work for that. Also, `make
> run` will start a locally served test page at http://localhost:8000/
> <http://h5ks.mjt.lu/lnk/AEAAHpZTiiEAAAAAAAAAAGgH3uUAASJ-8QAAAAAAAAgArwBYqLzevEDLj7I6TJaVrbRJp9rrAgAHnU0/3/_z26ESO4hE7rs-_6pB2pCw/aHR0cDovL2xvY2FsaG9zdDo4MDAwLw>
>
> Romain
>
> 2017-02-17 22:50 GMT-06:00 Mark E <[email protected]>:
>
>>
>> Here's where I am at on this:
>>
>> After the WebRTC caller connects to the webcaster radio host (which
>> works fine, the show host and caller can talk to each other) then I do
>> this (using the webcaster's current AudioContext - so I think anyway -
>> where event.stream is the caller's audio stream object (taking WebRTC
>> audio into Web Audio for processing, which is definitely possible) :
>>
>> source = Webcaster.node.context.createMediaStreamSource(
>> event.stream );
>> gain_node = Webcaster.node.context.createGain();
>> source.connect( gain_node );
>>
>> // So with the above I have a gain node with the caller's audio stream
>> connected to it, or so I think.
>> // Now I connect the gainNode to the Webcaster's node.context.destination
>>
>> gain_node.connect( Webcaster.node.context.destination );
>>
>> No errors are returned in the JS console, but the code doesn't cause the
>> caller's audio stream to be sent out via webcaster to liquidsoap and on
>> to Icecast (I'm monitoring the audio stream in a different computer via
>> a MP3 stream player connected to the Icecast stream). The webcaster
>> host's audio is sent through liquidsoap to Icecast, I can hear that.
>>
>> What am I doing wrong?
>>
>> Mark
>>
>> On 02/17/2017 08:08 AM, Mark E wrote:
>> > Hi,
>> >
>> > First let me say that webcaster is a great tool - thank you very much
>> > for creating it along with webcast.js and liquidsoap.
>> >
>> > I would like to integrate a simple WebRTC audio-only chat into the
>> > webcaster client code so that an inbound WebRTC caller is connected to
>> > the webcaster host's (the person operating webcaster) voice audio and
>> > mixed into the outbound audio stream being sent to liquidsoap.
>> >
>> > I have the chat built and it works fine. Now I need to integrate that
>> > into the webcaster app. I need some advice please. This is what I think
>> > should be done:
>> >
>> > - When the "Start streaming" button is clicked and the app does its
>> > navigator.mediaDevices.getUserMedia() call to get permission for the
>> mic
>> > and the result stream, add code to connect to my WebRTC signaling
>> server..
>> >
>> > - On a different web page where people listen to the stream, add a
>> > button to initiate a call to the studio. When the button is clicked the
>> > listener is asked for permission to use the mic and then on success is
>> > connected to the signaling server.
>> >
>> > - Signaling server sends an offer to connect to the webcaster host,
>> > which then (after the usual WebRTC negotiation) results in a RTC peer
>> > connection between the show and the caller.
>> >
>> > - At that point I need to somehow connect the caller's audio stream
>> > (object) into the overall webcaster audio mix so that the conversation
>> > goes out to liquidsoap as usual.
>> >
>> > The question I have is where and how within the webcaster client.js do I
>> > connect the callers stream to the outbound audio stream? I'm not an
>> > expert with backbone and underscore (used with webcaster) but I can wing
>> > it as I learn if someone can please point me to the right things to do.
>> >
>> > Thanks for you time in considering this!
>> >
>> > Mark
>> >
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